Several conference papers and patents address the issue of decreasing the protocol overhead of the transmission of small sized voice packets in IP based packet networks by proposing multiplexing solutions.
According to the known solution, for example, document US 2006/0120347 A1 proposes multiplexing in the transport layer for general IP based networks and document US2006/0133372 A1 proposes similar UDP based multiplexing schemes for mobile networks. The multiplexing can also be executed in the IP protocol of the network layer as proposed by document U.S. Pat. No. 6,920,125 B1 for mobile networks and by document “Transport Multiplexing Protocol (TMux)” by P. Cameron et al, RFC 1692, August 1994, for general IP based networks. Further, document “RTP payload multiplexing between IP telephony gateways” by B. Subbiah et al, Globecom '99, pp. 1121-1127, Rio de Janeiro 1999, proposes a multiplexing solution operating in the RTP layer for IP telephony gateways.
These solutions are common in the sense that they propose non-adaptive multiplexing operation, i.e., the number of maximum multiplexed packets are determined on a-priori given thresholds (waiting time/number of packets threshold) and the actual network state does not influence the operation.
Further, adaptive multiplexing solutions for the adaptive setting of multiplexing parameters are proposed in documents US 2009/0103504 A1, document “Adaptive Multiplexing Based on E-model for Reducing Network Overhead in Voice over IP Security Ensuring Conversation Quality” by R. M. Pereira et al, 2009 Fourth International Conference on Digital Telecommunications, pp. 53-58, Colmar, France, 2009, ISBN: 978-0-7695-3695-8, document “Measurement-Based Multi-Call Voice Frame Grouping in Internet Telephony” by H. Kim et al, IEEE Communications Letters, vol. 6, no. 5, May 2002, and document “Voice-TFCC: A TCP-Friendly Congestion Control Scheme for VoIP Flows” by A. Trad et al, PIMRC 2008, 15-18 Sep. 2008, Cannes, France. Pereira, Kim and Trad consider multiplexing in a general IP based network while document US 2009/0103504 A1 assumes an LTE communication network. Document US 2009/0103504 A1 proposes to set the maximum number of multiplexed packets according to the remaining storage capacity of the reception buffer. This solution uses a static timer value, thus the delay added by the multiplexing cannot be adapted to the actual network state.
Pereira proposes to set the multiplexing timer value according to the quality of the voice calls. This needs the evaluation of the voice calls individually. Due to complexity concerns this solution is hardly applicable in telecommunication systems.
According to Kim, the value of the multiplex timer is determined from the measured mouth to ear delay. The delay is predicted from the measurements using the retransmission timeout calculation algorithm of the Transport Control Protocol (TCP) in order to temper the variation of the predicted mouth to ear delay. In a mobile communication network the significant part of the mouth to ear delay can be the air interface delay, thus the application of this solution would need the measurement of the delay of each connection that is not feasible due to complexity issues.
Moreover, Trad proposes to set the number of multiplexed voice calls according to a TCP friendly rate setting formula. This solution varies only the number of multiplexed voice calls and uses a fixed multiplex timer, thus it cannot minimize the added delay.
Hence, the above mentioned solutions are operating based on the information from the receiving entity on the quality of the connection. In contrast thereto, the solution according to embodiments of the present invention to be described later operate based only on information that is available at the multiplexer entity side.